can someone please direct me to some good documentation for setting up IP trunking on an Avaya S8710 CM 3.o, just basic admin as far as what to program (signal group tye h.323?)
ok have everything programmed between 2 PBX' but when i try to place a call from one to the error i fail.
display trunk-group 201 Page 1 of 19
TRUNK GROUP
Group Number: 201 Group Type: isdn CDR Reports: y
Group Name: OUTSIDE CALL COR: 1 TN: 1 TAC: 5400
Direction: two-way Outgoing Display? n Carrier Medium: IP
Dial Access? n Busy Threshold: 255 Night Service:
Queue Length: 0
Service Type: tie Auth Code? n TestCall ITC: unre
Far End Test Line No:
TestCall BCC: 0
TRUNK PARAMETERS
Codeset to Send Display: 0 Codeset to Send National IEs: 6
Max Message Size to Send: 260 Charge Advice: none
Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc
Trunk Hunt: cyclical QSIG Value-Added? n
Digital Loss Group: 18
Incoming Calling Number - Delete: Insert: Format:
Bit Rate: 1200 Synchronization: async Duplex: full
Disconnect Supervision - In? y Out? n
Answer Supervision Timeout: 0
display trunk-group 201 Page 3 of 19
TRUNK GROUP
Administered Members (min/max): 1/3
GROUP MEMBER ASSIGNMENTS Total Administered Members: 3
Port Code Sfx Name Night Sig Grp
1: T00296 Test 201
2: T00297 201
3: T00298 201
display signaling-group 201 Page 1 of 5
SIGNALING GROUP
Group Number: 201 Group Type: h.323
Remote Office? n Max number of NCA TSC: 0
SBS? n Max number of CA TSC: 0
IP Video? n Trunk Group for NCA TSC:
Trunk Group for Channel Selection: 201
Supplementary Service Protocol: a Network Call Transfer? n
T303 Timer(sec): 10
Near-end Node Name: clan05a10 Far-end Node Name: clan2a18_nr15
Near-end Listen Port: 1720 Far-end Listen Port: 1720
Far-end Network Region: 15
LRQ Required? n Calls Share IP Signaling Connection? y
RRQ Required? n
Media Encryption? n Bypass If IP Threshold Exceeded? n
DTMF over IP: out-of-band Direct IP-IP Audio Connections? y
IP Audio Hairpinning? y
Interworking Message: PROGress
DCP/Analog Bearer Capability: 3.1kHz
Yes, the call will use any avaiable medpro in the same network region. If you have 2 medpros in the same region calls will load balance between them. First call to medpro 1, second call to medpro 2, third call to medpro 1, etc...
That’s what I thought, Thanks for the info. Couple more questions: Is there a way to real time status or monitor DSP resources? It would be nice to see what resources are in use for IP stations and trunks and what not. I found the list measurements command but it does not seem to be populating any info for DSP resources.
Also when you status a trunk group normally it gives you the physical location of the trunk (IE: 03E1102), however when you status a IP trunk group it give you something like 0009/001 T00140. What is that referencing?
I appreciate the information. I'm trying to catch up on the H.323 and SIP stuff. It seems we are moving that way quickly. I think guys like me are either going to learn it or look for a new job
You can do a 'status media-processor 01AXXXX' to see the status of your medpro board. This will show you how many calls active on that board at that point in time.
T00140 is an IP trunk, S00XXX would be an IP phone (hard or soft).
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